Processor for an FM signal receiver and processing method

ABSTRACT

A processing unit for processing a multi-channel audio signal has a delay element ( 40 ) for delaying an FM sum signal (sum) and a converter arrangement (T) for converting an FM difference signal (diff) and a noise signal (diffnoise) to the frequency domain. Frequency-based noise suppression is used to derive a de-noised frequency-domain difference signal using a gain function which is limited to a maximal and a minimal value. This is then converted to the time domain, and the first and second audio signals are calculated from a delayed sum signal (sum 2 ) and the de-noised time domain difference signal (diff 2 ).

CROSS-REFERENCE TO RELATED APPLICATIONS

This application claims the priority under 35 U.S.C. §119 of Europeanpatent application no. 12151208.1, filed on Jan. 16, 2012, the contentsof which are incorporated by reference herein.

SUMMARY OF THE INVENTION

This invention relates to an FM signal receiver and reception method.

In FM broadcasting, the demodulated FM-stereo signal consists of a monoaudio signal in the form of a sum signal (L+R, “main” channel),transmitted as baseband audio in the range of 30 Hz to 15 kHz, a pilottone of 19 kHz and a stereo difference signal (L-R, “Sub” channel)amplitude modulated on a 38 kHz sub carrier, occupying the range of 24kHz to 53 kHz.

The representation of a stereo audio signal as a sum and a differencesignal, rather than a left and a right audio signal, ensurescompatibility with mono receivers, which only use the main channel.

FIG. 1 is a schematic diagram of an FM tuner, which converts an FM radiobroadcast into audio signals. The tuner generates three output signalsfrom a given FM broadcast: the sum signal (“sum”), the difference signal(“diff”) and a signal that is representative of the noise that ispresent in the difference signal (“diffnoise”). The diffnoise signal isa signal with signal statistics that are similar to the noise componentpresent in the difference signal.

The noise information can for example be derived from the differencesignal diff. FIG. 2 a is a schematic representation of the PowerSpectral Density (PSD) of an input FM multiplex signal. The input signalcomprises a baseband sum signal 21 (between 0 and 15 kHz), a 19 kHzpilot tone 22 and a double sideband suppressed carrier modulateddifference signal 23 (between 23 and 53 kHz). A noise signal 24 is alsopresent, which tends to increase with increasing frequency.

The difference signal 23 is effectively available twice, once in thefrequency range from 23 to 38 kHz and once in the frequency range from38 to 53 kHz. Hence, using this knowledge the received differencesignal, which consists of the original difference signal plus theadditional noise component is obtained, but an approximation of thenoise signal diffnoise can also be derived.

The two signals can be obtained as illustrated in FIGS. 2 b and 2 c.Quadrature modulation (modulation with complex-exponential) is firstapplied to the original input spectrum of FIG. 2 a with a modulationfrequency of 38 kHz. This results in a complex valued signal having thespectrum indicated in FIG. 2 b. This signal is then low pass filtered toapproximately 15 kHz, resulting in the signal shown in FIG. 2 c (theband pass filter is indicated by the band pass function 27). Theresulting complex valued signal comprises the demodulated receivedsignal diff as well as the complex-modulated signal diffnoise. By takingthe real part and imaginary part of this signal, the total receiveddifference signal (i.e. including the received noise) and the noisesignal diffnoise can be obtained.

The difference signal is real-valued, leading to a symmetrical spectrum.The noise does not have a symmetrical spectrum (due to demodulation).The part of the total spectrum that is symmetrical ends up in thereal-valued part of the signal, and the complex-valued part containsmuch of the noise, although part of the noise will also end up in thereal-valued part.

As a consequence, a ratio of the signal plus noise to the noise (SNNR)of the difference signal can be estimated. The power of the differencesignal consists of the power of the difference signal plus the power ofthe noise estimate, under the assumption that there is zero correlationbetween the difference signal and the positive and negative noisecomponents.

Referring back to FIG. 1, the left and right audio signals arereconstructed from the sum and difference signals by, respectivelyadding and subtracting the sum and difference signals (possibly with ascaling factor). This process is often referred to as dematrixing(labeled “DM” in FIG. 1).

When the reception quality of the FM tuner deteriorates, noise degradesthe FM signal. However, the sum and difference signals are influenceddifferently.

When the received FM signal contains white noise, the correspondingdemodulated noise component linearly increases with frequency. Since theMain channel signal is present in the low frequency area (up to 15 kHz),the signal-to-noise ratio (SNR) is considerably better in the sum signalthan in the difference signal.

This means that in noisy conditions, the mono signal (only the sumsignal) contains less noise than the stereo signal (since the left andright signals are derived from the sum and the difference signal). Thereexist approaches that exploit the fact that the sum signal is lessaffected by the noise than the difference signal, by switching from astereo to a mono signal depending on the FM reception quality or othermeasures related to the expected SNR of the difference signal, e.g.,using the signal power of the diffnoise signal.

In EP 0 955 732, an approach for de-noising the difference signal isdescribed which divides the difference signal into sub-band signals,which are individually multiplied with factors to generate a frequencyselective weighted stereo difference signal. The factors can be set as afunction of the signal quality of the RF-signal, the signal energy inthe sum signal or the signal energy in the difference signal. A furtherextension is described in U.S. Pat. No. 7,110,549 which takes intoaccount perceptual masking effects in the computation of the weightingfactors.

A different approach is disclosed in US2011/0235809. A difference signalis synthesised as a weighted sum of the original difference signal and ade-correlated version of the sum signal (also per sub-band). Theweighting factors depend on the cross-correlation between sum anddifference signals, and the powers of the sum and difference signals.

In WO 2008/087577, a portion of the sum signal is added to thesynthesised difference signal.

According to the invention, there is provided a processing unit and amethod as defined in the independent claims.

In one aspect, the invention provides a processing unit for processing amulti-channel audio signal comprising:

-   -   a delay element for delaying an FM sum signal (sum) representing        a sum of a first and a second audio signal;    -   a converter arrangement for converting an FM difference signal        (diff) representing a difference between the first and second        audio signals to the frequency domain, and for converting a        noise signal (diffnoise) representing noise in the FM difference        signal to the frequency domain;    -   a frequency-based noise suppression unit for suppressing noise        in the difference signal (diff) to derive a de-noised        frequency-domain difference signal on the basis of a gain        function based at least on the frequency-domain noise signal        amplitude, wherein the gain function is limited to a maximal and        a minimal value;    -   an inverse transform unit for inversely transforming the        de-noised difference signal or a signal derived from it to the        time domain; and    -   a de-matrixing unit for computing the first and second audio        signals from the delayed sum signal and time domain difference        signal.

The invention applies a frequency-domain noise suppression technique tothe difference signal, before it is recombined with the sum signal. Theenhanced (de-noised) difference signal is mixed with the sum signal andthis masks many of the problems (artifacts) that would be audible whenlistening to the difference signal only.

In a preferred example, a gain unit is provided for applying additionalgain factors to the de-noised frequency-domain difference signal toderive a processed frequency-domain difference signal, and the inversetransform unit is for inversely transforming the processedfrequency-domain difference signal.

In this way, the frequency-domain de-noising is combined with adaptivegain factors to ensure that the noise level in the output signal issufficiently low. The gain factors can adapt slowly over time inresponse to noise changes.

Traditional frequency-domain noise suppression algorithms compute a gainfunction that is a function of the amplitude spectra of the originalsignal and of an estimated disturbance. In this case, the originalsignal is the difference signal (diff) and the disturbance is the noisesignal (diffnoise). This diffnoise signal is either available from theFM tuner as outlined above or it can be estimated using techniques thatare well known (such as noise floor estimation).

To avoid typical frequency-domain noise suppression artifacts, such asmusical noise, the amount of noise suppression is limited. The adaptivegain factors are controlled in such a way that the combination of thenoise suppression and the adaptive gain factors yield an audio signal inwhich the noise level is sufficiently low.

The processing unit preferably comprises a controller for controllingthe gain unit. The adaptive gain factors can be based on a residualnoise level. The converter arrangement can also be used for convertingthe sum signal to the frequency domain, and the adaptive gain factorsthen can be based on a residual noise level in combination with thefrequency-domain sum signal.

The gain function and adaptive gain factors can be controlled such thatthe expected signal-to-noise ratio in the first and second audio signalsmeet predefined criteria. The criteria can be based on the ratio of theexpected peak value in the first audio signal, the second audio signalor the sum signal, and the noise level in the processed differencesignal.

The invention also provides a receiver comprising an FM tuner and aprocessing unit of the invention.

Another aspect of the invention provides a method of processing amulti-channel audio signal comprising:

-   -   delaying an FM sum signal (sum) representing a sum of a first        and a second audio signal;    -   converting an FM difference signal (diff) representing a        difference between the first and second audio signals to the        frequency domain, and converting a noise signal (diffnoise)        representing noise in the FM difference signal to the frequency        domain;    -   suppressing noise in the frequency-domain difference signal        using a frequency-based noise suppression unit to derive a        de-noised difference signal on the basis of a gain function        based on at least the frequency-domain noise signal amplitude,        wherein the gain function is limited to a maximal and a minimal        value,    -   inversely transforming the de-noised frequency-domain difference        signal or a signal derived from it to the time domain; and    -   computing the first and second audio signals from the delayed        sum signal (sum2) and time domain difference signal.

BRIEF DESCRIPTION OF THE DRAWINGS

Examples of the invention will now be described in detail with referenceto the accompanying drawings, in which:

FIG. 1 shows the signals obtained during FM signal reception;

FIG. 2 (consisting of FIGS. 2A, 2B and 2C), shows how a noise estimatecan be obtained;

FIG. 3 is a flowchart showing the method of the invention;

FIG. 4 is a first example of receiver of the invention; and

FIG. 5 is a second example of receiver of the invention.

DETAILED DESCRIPTION OF THE EMBODIMENTS

The invention provides a processing unit for processing a multi-channelaudio signal in which frequency-based noise suppression is applied tothe difference signal. This de-noising can be combined with adaptivegain factors which are based at least on a frequency-domain noise signalamplitude. The noise suppression gain function is limited to a maximaland a minimal value, to reduce artifacts caused by the noisesuppression.

FIG. 3 shows a flowchart of the invention, in which the optionalcomponents are represented by dashed lines. The process starts from thethree time-domain signals: sum, diff and diffnoise (the outputs of theFM tuner, see FIG. 1).

The diff and diffnoise signals are transformed to the frequency domainin step 30. A gain function is computed in step 31 on the basis of thesignal amplitudes and this gain function is limited to a maximal and aminimal value in step 32. The gain function is applied to thefrequency-domain difference signal diff in step 33.

This gain function implements a limited noise suppression function inthe frequency domain.

Additional (adaptive) gain factors (a single one or one per frequency)are computed in step 34 either on the basis of the residual noise level,or of the residual noise level in combination with the frequency-domainsum signal and these gain factors are also applied in step 35 to thedifference signal. If a frequency-domain sum signal is used, there is atransformation step shown as 38 to convert from the time domain to thefrequency domain.

Finally, the processed difference signal is inversely transformed to thetime domain in step 36, and the stereo signal is computed from thedelayed sum (the delay step shown as 39) and processed diff signal. Thisis the dematrixing operation shown as step 37.

Traditional frequency-domain noise suppression algorithms compute a gainfunction that is a function of the amplitude spectra of the originalsignal and of the (estimated) disturbance. In the current context, theoriginal signal is the stereo difference signal (diff) and thedisturbance is the diffnoise signal. This diffnoise signal is eitheravailable from the FM tuner or it can be estimated using techniques thatare well known.

It is well known that the enhanced signal, i.e., the de-noised signalafter noise suppression, may contain audible artifacts, such as musicalnoise, after processing with the traditional frequency-based techniques.Musical noise is an effect which is artificially produced by spectralsubtraction, and manifests itself as a tin-like sound. Musical noise canbe removed by post-processing but this adds further complexity to theprocessing.

To alleviate this problem, the amount of noise suppression can belimited to a fixed amount by imposing a minimal and/or a maximal valueto the de-noising gain function. In this way, there may be some residualnoise after noise suppression, but the artifacts are reduced.

In the context of FM stereo signals, the audio signal can be enhanced inseveral ways, assuming that the sum signal is less noisy than thedifference signal:

-   -   apply noise suppression to the sum signal;    -   apply noise suppression to the difference signal;    -   attenuate the difference signal by a certain factor, such that        the resulting left and right signals contain less of the        difference signal, and thus, less of the noise present in the        difference signal.

The first two possibilities do not alter the stereo image, but the thirdone does: by attenuating the difference signal, the stereo image willbecome less wide (it will tend more towards mono).

In the preferred example, the invention processes the difference signalusing a combination of a traditional frequency-based noise suppressionmethod (with gain limitation) and adaptive gain factors to furtherreduce the noise in the left and right audio channels if necessary.

Frequency-based noise suppression usually computes a gain function,H₁(ω) where ω denotes frequency, which is applied to the signal thatneeds to be enhanced. It is typically a function of the amplitudespectra of the input signal and of the disturbance. As an example, thefollowing gain function can be used (for more details, reference is madeto Loizou, 2007, chapter 5, “Speech Enhancement; Theory and Practice”1st edition RCR press):

$\begin{matrix}{{{H_{1}(\omega)} = \sqrt{\frac{{{D(\omega)}}^{2} - {\alpha{{N(\omega)}}^{2}}}{{{D(\omega)}}^{2}}}},} & (1)\end{matrix}$where |N(ω)| and |D(ω)| are the magnitudes of respectively, thediffnoise and the diff signal at frequency ω, and α is anover-subtraction factor.

This gain function has a gain of 1 or less (i.e. it providesattenuation) and the level of attenuation at each frequency is afunction of the noise level at that frequency.

The gain function can be limited to a certain minimal value, H_(min)(ω),and a certain maximal value, H_(max)(ω), to reduce audible artifacts dueto the noise suppression:H ₂(ω)=min{max{H ₁(ω),H _(min)(ω)},H _(max)(ω)}  (2)

The minimum and maximum values are functions of frequency, as shown.

This gain function may further be smoothed (both across time and acrossfrequency), possibly using asymmetrical smoothing time constants forgain increases and decreases to further reduce artifacts. The enhanced(i.e. de-noised) difference signal is obtained by applying the gain tothe original difference signal:D1(ω)=H ₂(ω)D(ω)  (3)

Additional (frequency dependent) gain factors are applied to thisde-noised difference signal:D2(ω)=γ(ω)H ₂(ω)D(ω)  (4)

where γ(ω) is the additional (adaptive) gain factor with respect tofrequency.

When the gain is limited to a minimal value H_(min)(ω), the enhanceddifference signal, will have a residual noise level, which can beestimated as:N1(ω)=N(ω)(H ₂(ω)−H ₁(ω))  (5)

Various different gain functions can be applied. These are developed inorder to provide a good estimate of the enhanced signal, and thedifferent types depend on the different assumptions and simplificationstaken. Another often-used variant is the MMSE (minimalmean-square-error) gain function, which requires the a priori signal tonoise ratio (SNR), the estimation method of which leads to differentspecific MMSE functions.

By imposing a maximal and/or minimal value on the de-noising gainfunction, the dynamic range of the gain function is limited. This way, atrade-off can be made between the amount of noise suppression and thelevel of (possibly audible) artifacts, since many artifacts are causedby large abrupt changes in the gain function. Keeping Hmax at unity,Hmin can be increased from 0 to unity, corresponding to solutions withmaximal noise suppression and most risk of audible artifacts (Hmin=0) tosolutions with no noise suppression and no artifacts (Hmin=1). A similarreasoning holds for Hmax. These values can therefore be set according tothe desired result.

A first embodiment is shown in FIG. 4. The system has three inputs,namely the sum signal, the diff signal and the diffnoise signal. The sumsignal is delayed by a delay element 40. The modules labeled “T” and“T⁻¹” denote a frequency transform and its inverse. This can be an FFTand inverse FFT (including windowing and overlap-add techniques), butalso filter bank type of transforms, such as quadrature mirror filterbanks (QMF) or others.

The noise suppression module 41 (NS) computes a gain function andapplies it to the frequency-domain difference signal D(ω). The output ofthe module 41, D1(ω), is multiplied by a gain factor, γ(ω), yieldingD2(ω), after which the inverse transform is computed.

The output of the noise suppression module can be considered to be afrequency-domain de-noised difference signal D1(ω). The output after theadaptive gain element 42 can be considered as a processedfrequency-domain difference signal D2(ω). After the inverse transform, aprocessed time domain difference signal “diff2” results.

The left and right signals are obtained in the dematrixing module “DM”from the delayed sum signal, sum2, and the output signal of the inversetransform unit, diff2.

The gain factors, γ(ω), are controlled by a control module 44 (ctrl)which adapts the gain factors in such a way that the expected noiselevel in the difference signal, e.g., estimated using N(ω) (Eq. (4)),complies to a certain criterion. This criterion can be based on theratio of the expected peak value of the original difference signal andthe residual noise level in the enhanced difference signal (perfrequency bin), or it can be based on a perceptually motivatedcriterion, e.g., using perceptual masking thresholds or partialloudness.

Setting a predefined peak-to-residual-noise-level to a low value may betoo strict in many cases: the residual noise may be masked by signalpower in neighbouring frequency regions. This masking effect is takeninto account when a predefined signal-to-mask ratio is used ascriterion, which is the ratio between residual noise and the perceptualmasking threshold (a 0 dB ratio corresponds to the limit at which thenoise is inaudible). Partial loudness can be used to predict theperceived loudness of the residual noise in the presence of othersources (the broadcast, but also, e.g., ambient noise).

The adaptation of the gain factors should be slow (for example timeconstants between 100 ms and several seconds) so as not to generateaudible artifacts.

A second embodiment is shown in FIG. 5. The sum signal sum is alsotransformed to the frequency domain, and it constitutes an additionalinput to the control module 44. The gain factors are now adapted in sucha way that the expected signal-to-noise ratio in the left and rightaudio channel complies to a certain criterion, as well as or instead ofthe difference signal as in the first embodiment.

This criterion can be based on the ratio of the expected peak value inthe left audio, right audio or sum signal, and the noise level in thede-noised difference signal (e.g., computed on the basis of N(ω)), or itcan be based on a perceptually motivated criterion, e.g., usingperceptual masking thresholds or partial loudness.

The second embodiment provides a control mechanism that better reflectsthe final stereo signal. Indeed, a criterion based on the left and rightoutput signals is used, rather than only of the output differencesignal. Artifacts or residual noise that are audible in the outputdifference signal are not necessarily audible after dematrixing, sincethey can be masked by the sum signal.

The invention can be implemented as a software module that processes anFM stereo signal. Thus, the invention also relates to computer softwarethat can be run by a processor forming part of an FM receiver.

The input signals are the sum signal (‘sum’), the difference signal(‘diff’) and a noise signal that has similar signal characteristics asthe noise component in the difference signal (diffnoise). The modulerequires the following components:

-   -   a frequency transform.    -   a frequency-domain noise suppression module that enhances the        difference signal.    -   an adaptive gain unit that attenuates the frequency-domain        enhanced difference signal.    -   a de-mixing matrix to generate the left and right audio signal        from the sum signal and the enhanced difference signal.

The invention can be part of an FM stereo tuner. It can be implementedas a software module to improve signal quality when the FM receptionquality deteriorates.

The preferred implementation of the invention combines frequency domainde-noising with adaptive gain factors applied to the difference signal.However, advantages are obtained when only the frequency-domainde-noising is applied to the difference signal, and the additional useof adaptive gain factors is therefore preferred but optional.

Other variations to the disclosed embodiments can be understood andeffected by those skilled in the art in practicing the claimedinvention, from a study of the drawings, the disclosure, and theappended claims. In the claims, the word “comprising” does not excludeother elements or steps, and the indefinite article “a” or “an” does notexclude a plurality. A single processor or other unit may fulfill thefunctions of several items recited in the claims. The mere fact thatcertain measures are recited in mutually different dependent claims doesnot indicate that a combination of these measured cannot be used toadvantage. A computer program may be stored/distributed on a suitablemedium, such as an optical storage medium or a solid-state mediumsupplied together with or as part of other hardware, but may also bedistributed in other forms, such as via the Internet or other wired orwireless telecommunication systems. Any reference signs in the claimsshould not be construed as limiting the scope.

The invention claimed is:
 1. A processing unit for processing amulti-channel audio signal, comprising: a delay element configured todelay an FM sum signal representing a sum of a first and a second audiosignal; a converter configured to convert an FM difference signalrepresenting a difference between the first and second audio signals tofrequency domain, and configured to convert a noise signal representingnoise in the FM difference signal to the frequency domain; afrequency-based noise suppression unit configured to suppress noise inthe difference signal to derive a de-noised frequency-domain differencesignal based on a gain function based at least on an amplitude of thefrequency-domain noise signal, wherein the gain function is limited to amaximal and a minimal value; an inverse transform unit configured toinversely transform the de-noised frequency-domain difference signal ora signal derived from the de-noised frequency-domain difference signalto time domain; and a de-matrixing unit configured to compute the firstand second audio signals from the delayed FM sum signal and time domaindifference signal; and further comprising a gain unit configured toapply additional gain factors to the de-noised frequency-domaindifference signal to derive a processed frequency-domain differencesignal, wherein the inverse transform unit is configured to inverselytransform the processed frequency-domain difference signal.
 2. Aprocessing unit as claimed in claim 1, further comprising a controllerconfigured to control the gain unit.
 3. A processing unit as claimed inclaim 2, wherein the additional gain factors are based on a residualnoise level.
 4. A processing unit as claimed in claim 2, wherein theconverter arrangement is configured to convert the sum signal to thefrequency domain, and wherein the additional gain factors are based on aresidual noise level in combination with the frequency-domain sumsignal.
 5. A processing unit as claimed in claim 4, wherein the gainfunction and the additional gain factors are controlled such that anexpected signal-to-noise ratio in the first and second audio signalsmeet predefined criteria.
 6. A processing unit as claimed in claim 5,wherein the criteria are based on a ratio of the expected peak value inthe first audio signal, the second audio signal or the sum signal, andthe noise level in the processed time-domain or frequency-domaindifference signal.
 7. A receiver comprising an FM tuner and a processingunit as claimed in claim
 1. 8. A method of processing a multi-channelaudio signal, comprising: delaying an FM sum signal representing a sumof a first and a second audio signal; converting an FM difference signalrepresenting a difference between the first and second audio signals toa frequency domain, and converting a noise signal representing noise inthe FM difference signal to the frequency domain; suppressing noise inthe frequency-domain difference signal using a frequency-based noisesuppression unit to derive a de-noised difference signal based on a gainfunction based on at least the frequency-domain noise signal amplitude,wherein the gain function is limited to a maximal and a minimal value,inversely transforming the de-noised frequency-domain difference signalor a signal derived from the de-noised frequency-domain differencesignal to the time domain; and computing the first and second audiosignals from the delayed sum signal and time domain difference signal;and further comprising applying additional gain factors to the de-noisedfrequency-domain difference signal to derive a processedfrequency-domain difference signal, wherein the inverse transforming isapplied to the processed frequency-domain difference signal.
 9. A methodas claimed in claim 8, wherein the additional gain factors are based ona residual noise level.
 10. A method as claimed in claim 8, furthercomprising converting the sum signal to the frequency domain, and theadditional gain factors are based on a residual noise level incombination with the frequency-domain sum signal.
 11. A method asclaimed in claim 10, wherein the gain function and the additional gainfactors are controlled such that the expected signal-to-noise ratio inthe first and second audio signals meet predefined criteria.
 12. Amethod as claimed in claim 11, wherein the criteria are based on theratio of the expected peak value in the first audio signal, the secondaudio signal or the sum signal, and the noise level in the processedfrequency-domain or time-domain difference signal.
 13. A non-transitorycomputer-readable storage medium containing a program configured toimplement the method of claim 8, when said program is run on a computer.